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webrtcbin: support using an a=mid value from the sink/transceiver caps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
This commit is contained in:
Matthew Waters
GStreamer Marge Bot
parent
400faf7361
commit
9a758d78a9
2 changed files with 129 additions and 22 deletions
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@ -2974,7 +2974,7 @@ sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media,
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* multiple dtls fingerprints https://tools.ietf.org/html/draft-ietf-mmusic-4572-update-05
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*/
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GstSDPMessage *last_offer = _get_latest_self_generated_sdp (webrtc);
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gchar *direction, *sdp_mid, *ufrag, *pwd;
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gchar *direction, *ufrag, *pwd, *mid;
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gboolean bundle_only;
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GstCaps *caps;
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GstStructure *extmap;
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@ -3156,17 +3156,34 @@ sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media,
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/* Some identifier; we also add the media name to it so it's identifiable */
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if (trans->mid) {
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gst_sdp_media_add_attribute (media, "mid", trans->mid);
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mid = g_strdup (trans->mid);
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} else {
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/* Make sure to avoid mid collisions */
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while (TRUE) {
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sdp_mid = g_strdup_printf ("%s%u", gst_sdp_media_get_media (media),
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webrtc->priv->media_counter++);
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if (g_hash_table_contains (all_mids, (gpointer) sdp_mid)) {
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g_free (sdp_mid);
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} else {
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gst_sdp_media_add_attribute (media, "mid", sdp_mid);
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g_hash_table_insert (all_mids, sdp_mid, NULL);
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break;
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const GstStructure *s = gst_caps_get_structure (caps, 0);
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mid = g_strdup (gst_structure_get_string (s, "a-mid"));
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if (mid) {
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if (g_hash_table_contains (all_mids, (gpointer) mid)) {
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g_set_error (error, GST_WEBRTC_ERROR,
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GST_WEBRTC_ERROR_INTERNAL_FAILURE,
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"Cannot re-use mid \'%s\' from the caps in m= line %u that has "
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"already been used for a previous m= line in the SDP", mid,
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media_idx);
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return FALSE;
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}
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g_hash_table_insert (all_mids, g_strdup (mid), NULL);
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} else {
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/* Make sure to avoid mid collisions */
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while (TRUE) {
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mid = g_strdup_printf ("%s%u", gst_sdp_media_get_media (media),
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webrtc->priv->media_counter++);
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if (g_hash_table_contains (all_mids, (gpointer) mid)) {
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g_free (mid);
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} else {
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gst_sdp_media_add_attribute (media, "mid", mid);
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g_hash_table_insert (all_mids, g_strdup (mid), NULL);
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break;
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}
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}
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}
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}
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@ -3190,12 +3207,12 @@ sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media,
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}
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if (bundled_mids) {
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const gchar *mid = gst_sdp_media_get_attribute_val (media, "mid");
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g_assert (mid);
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g_string_append_printf (bundled_mids, " %s", mid);
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}
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g_clear_pointer (&mid, g_free);
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gst_caps_unref (caps);
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return TRUE;
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@ -4155,12 +4172,13 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
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rtp_trans->mline, rtp_trans->kind,
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webrtc_kind_from_caps (answer_caps));
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if (!trans->do_nack) {
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answer_caps = gst_caps_make_writable (answer_caps);
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for (k = 0; k < gst_caps_get_size (answer_caps); k++) {
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GstStructure *s = gst_caps_get_structure (answer_caps, k);
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answer_caps = gst_caps_make_writable (answer_caps);
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for (k = 0; k < gst_caps_get_size (answer_caps); k++) {
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GstStructure *s = gst_caps_get_structure (answer_caps, k);
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/* taken from the offer sdp already and already intersected above */
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gst_structure_remove_field (s, "a-mid");
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if (!trans->do_nack)
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gst_structure_remove_fields (s, "rtcp-fb-nack", NULL);
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}
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}
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gst_sdp_media_set_media_from_caps (answer_caps, media);
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@ -1628,6 +1628,30 @@ GST_START_TEST (test_get_transceivers)
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GST_END_TEST;
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static void
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on_sdp_media_check_mid (struct test_webrtc *t, GstElement * element,
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GstWebRTCSessionDescription * desc, gpointer user_data)
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{
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const char **mid = user_data;
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guint i;
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for (i = 0; i < gst_sdp_message_medias_len (desc->sdp); i++) {
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const GstSDPMedia *media = gst_sdp_message_get_media (desc->sdp, i);
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gboolean seen_mid = FALSE;
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guint j;
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for (j = 0; j < gst_sdp_media_attributes_len (media); j++) {
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const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, j);
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if (g_strcmp0 (attr->key, "mid") == 0) {
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fail_unless (!seen_mid);
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seen_mid = TRUE;
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fail_unless_equals_string (attr->value, mid[i]);
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}
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}
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}
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}
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GST_START_TEST (test_add_recvonly_transceiver)
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{
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struct test_webrtc *t = test_webrtc_new ();
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@ -1640,11 +1664,12 @@ GST_START_TEST (test_add_recvonly_transceiver)
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media_format_count, &no_duplicate_payloads);
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VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (1),
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&media_formats);
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const char *expected_mid[] = { "gst", };
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VAL_SDP_INIT (mid, on_sdp_media_check_mid, expected_mid, &count);
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const gchar *expected_offer_setup[] = { "actpass", };
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VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &count);
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VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &mid);
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const gchar *expected_answer_setup[] = { "active", };
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VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
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&count);
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VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup, &mid);
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const gchar *expected_offer_direction[] = { "recvonly", };
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VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
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&offer_setup);
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@ -1661,7 +1686,7 @@ GST_START_TEST (test_add_recvonly_transceiver)
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t->on_pad_added = _pad_added_fakesink;
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/* setup recvonly transceiver */
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caps = gst_caps_from_string (OPUS_RTP_CAPS (96));
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caps = gst_caps_from_string (OPUS_RTP_CAPS (96) ", a-mid=(string)gst");
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direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY;
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g_signal_emit_by_name (t->webrtc1, "add-transceiver", direction, caps,
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&trans);
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@ -4425,6 +4450,69 @@ GST_START_TEST (test_renego_rtx)
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GST_END_TEST;
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GST_START_TEST (test_bundle_mid_header_extension)
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{
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struct test_webrtc *t = test_webrtc_new ();
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GstWebRTCRTPTransceiverDirection direction;
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GstWebRTCRTPTransceiver *trans;
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VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
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NULL, NULL);
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guint media_format_count[] = { 1, 1, };
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VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
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media_format_count, &no_duplicate_payloads);
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VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (1),
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&media_formats);
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const char *expected_mid[] = { "gst", };
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VAL_SDP_INIT (mid, on_sdp_media_check_mid, expected_mid, &count);
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const gchar *expected_offer_setup[] = { "actpass", };
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VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &mid);
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const gchar *expected_answer_setup[] = { "active", };
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VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup, &mid);
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const gchar *expected_offer_direction[] = { "recvonly", };
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VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
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&offer_setup);
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const gchar *expected_answer_direction[] = { "sendonly", };
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VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
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&answer_setup);
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GstCaps *caps;
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GstHarness *h;
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guint mline;
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char *trans_mid;
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/* add a transceiver that will only receive an opus stream and check that
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* the created offer is marked as recvonly */
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t->on_negotiation_needed = NULL;
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t->on_ice_candidate = NULL;
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t->on_pad_added = _pad_added_fakesink;
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/* setup recvonly transceiver */
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caps = gst_caps_from_string (OPUS_RTP_CAPS (96) ", a-mid=(string)gst");
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direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY;
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g_signal_emit_by_name (t->webrtc1, "add-transceiver", direction, caps,
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&trans);
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gst_caps_unref (caps);
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fail_unless (trans != NULL);
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g_object_get (trans, "mlineindex", &mline, NULL);
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fail_unless_equals_int (mline, -1);
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/* setup sendonly peer */
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h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
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add_fake_audio_src_harness (h, 96);
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t->harnesses = g_list_prepend (t->harnesses, h);
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test_validate_sdp (t, &offer, &answer);
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g_object_get (trans, "mlineindex", &mline, "mid", &trans_mid, NULL);
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fail_unless_equals_int (mline, 0);
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fail_unless_equals_string (trans_mid, "gst");
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g_clear_pointer (&trans_mid, g_free);
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gst_object_unref (trans);
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test_webrtc_free (t);
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}
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GST_END_TEST;
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static Suite *
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webrtcbin_suite (void)
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{
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@ -4480,6 +4568,7 @@ webrtcbin_suite (void)
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tcase_add_test (tc, test_codec_preferences_incompatible_extmaps);
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tcase_add_test (tc, test_codec_preferences_invalid_extmap);
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tcase_add_test (tc, test_renego_rtx);
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tcase_add_test (tc, test_bundle_mid_header_extension);
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if (sctpenc && sctpdec) {
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tcase_add_test (tc, test_data_channel_create);
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tcase_add_test (tc, test_data_channel_remote_notify);
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